[Asterisk-Users] "Hand-over" phone connections

[C F]

If you are using SIP then  you can do this by making sure that
canreinvite is set to yes in sip.conf for those 2 sip clients and:
1. No transcoding is taking place (you are using the same codec on both ends).
2. You don't have anything in the dial command that forces asterisk to
keep the stream (like t or T, etc.).
3. And that the 2 clients can reach each other nicely (if they are
both behind different NATs then there might be a problem, even if when
asterisk has no problem communictating with them).

On 11/5/05, Arik Funke <arik.funke at gmx.de> wrote:
> Hello,
>
> can somebody tell me if following is possible and if yes, how?
>
> Assume we have a Asterisk server to which two VoIP clients are connected
> over the internet (i.e. not internally). Now I would like to avoid
> having the connection run over the server but would like the server to
> tell the clients that they should contacts each other directly. Thus I
> would obviously avoid traffic, load on the server and reduce delays.
>
> Now how do I go about this? Thanks in advance for the help.
>
> Cheers,
> Arik
>
>
> PS: The clients have dynamic IP adresses and are possibly behind nat
> server...
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